Supporting Real-time Traffic: Preparing Your IP Network for ...
Supporting Real-time Traffic
Preparing Your IP Network for Video
January 5, 2006
Table of Contents
1 ..................................................................................................OVERVIEW 3
2 ................................................................................REAL TIME TRAFFIC 3
2.1 ............................................................. How is Real-Time Traffic Different? 3
3 ........................................... WHAT IS QUALITY OF SERVICE (QOS)? 5
3.1 ...................................................................... Network QoS Implementation 6
3.2 ................................................................................................ Classification 11
4 ............................................................................ BANDWIDTH DEMAND 12
5 .......................................................................AVAILABLE BANDWIDTH 14
6 ........................................................................DEMAND MANAGEMENT 15
7 .............................................. WAN VENDORS AND TECHNOLOGIES 17
7.1 .................................................................................. QoS in the WAN Core 17
7.2 ..................................................................... QoS in the WAN Access Links 17
7.3 .........................................................................................SLA Interpretation 17
7.4 ..................................................................Enterprise to WAN QoS Handoff 18
7.5 ........................................................... Real-Time over VPNs or the Internet 18
8 ..................................................................... NETWORK VERIFICATION 19
9 ........................................................................NETWORK MONITORING 20
10 .....................ENTERPRISE SERVICE LEVEL AGREEMENT (SLA) 21
11 .............................................................................................. CHECKLIST 23
12 ........................................................................................CONCLUSIONS 23
13 ..............................................................................................APPENDIX 1 24
14 .......................................................................................... REFERENCES 25
Voice over IP (VoIP) and video conferencing over IP create real-time traffic streams with
characteristics very different from run-of-the-mill data applications. If you are considering running
voice or video conferencing over your IP network, you should prepare for this different and often
challenging traffic type. The fact is that most enterprise networks are ill-equipped to carry video
conferencing traffic, and will need to be evaluated, tested, and possibly reconfigured and upgraded
to ensure acceptable quality.
Our goal is to identify the challenges posed by voice and video conferencing traffic, and provide a
blueprint to prepare your enterprise network to support them. In this paper we recommend
techniques for handling bandwidth, packet loss, jitter, latency and QoS implementation, as well as
testing methodologies for network verification and ongoing network monitoring. Lastly, we describe
an approach for specifying and managing the demands video conferencing places on IP network
An understanding of IP network design and deployment is helpful in understanding this guide, as is a
general knowledge of IP network deployment (switching, routing, bandwidths, error mechanisms,
etc.). Note that this paper does not address issues surrounding passing voice and video
conferencing traffic through Network Address Translating Routers (NATs), or the issues associated
2.0 Real Time Traffic
Real-time traffic supports real-time interactive applications, the most prominent of which are voice
and video conferencing. Both of these have users at each end of a connection who expect that what
they say or do will be transmitted ‘instantly’ to the other end of the connection, and the conversation
will proceed as if the two parties were in the same room. Some of the most difficult aspects of real-
time traffic come from this need for speed. We will see later how aspects of a normal data network
sometimes interfere with this requirement.
When we refer to real-time traffic in this document, it applies to voice over IP (VoIP) traffic, as well as
to both the video and audio streams of a video conferencing application.
2.1 How is Real-Time Traffic Different?
The Internet Protocol (IP) is at the heart of all modern networks, and is responsible for connecting
one endpoint to another. But by design, IP is an unreliable protocol. This means it is not designed
to insure that all packets sent from one endpoint arrive successfully at the other endpoint.
Data applications require a reliable transport, one that will insure all the bits of the data being sent
arrive successfully and correctly at the destination computer. To insure this result, our networks use
Transmission Control Protocol (TCP) on top of IP (TCP/IP). TCP insures each packet that is sent
arrives at the other end, and will send a packet again if one is lost. TCP then verifies all the data is
correct before delivering it to the application.
Data applications tend to be very bursty in their utilization of the network bandwidth. When a file or
block of data is ready to be transferred across the network, the sender wants to send the data as
quickly as possible, and then move on to other tasks. The result is very short bursts of activity on
the network, followed by periods of relatively low or no use of the network.
0 100 200 300 400 500 600 700 800 900 1000
Figure 1 - Typical Data Bandwidth Utilization
Figure 1 is a graph of a typical data application, showing periods of low and high network utilization.
IP-based networks often experience packet loss, and this is a normal part of the network operations.
In fact the TCP protocol uses packet loss as its flow control mechanism. TCP determines how
quickly to send data by increasing its send rate until packet loss occurs, and then backing off. This
occurs over and over for each TCP stream in the network.
Real-time traffic has very different characteristics. Real-time traffic results from a codec which is
sampling a continuous real-world environment (speech or images), and transmitting constant
updates of this information to reproduce a visual or auditory result. So the bandwidth utilization of
voice and video is constant during the time the application is running. Figure 2 is a graph of a 384K
video conference, showing both the audio and video streams, and their relatively constant use of
bandwidth during operation.
0 30 60 90 120 150 180 210 240 270 300
Figure 2 - Video Conferencing Bandwidth Utilization
A second characteristic of real-time streams is their sensitivity to delay. Because a real-time stream
is sampling and reproducing a continuous event, such as speech, individual data samples must
arrive at the destination end to be ‘played’ at the right time. If a packet is late, or is lost in transit,
then there will be a gap in the information available to the player, and the quality of the audio or
video reproduction will degrade. This degradation is significant, and occurs at relatively low levels of
packet delay and loss.
Because real-time packets must arrive in a timely manner, it is not possible for the transport protocol
to ask for a lost packet to be resent, and then wait for the source to try it again. The round trip delay
to the source and back again is too long, and the packet will have missed its play window. Because
TCP adds no value to these streams, they are carried instead with the User Datagram Protocol
(UDP), which has no recovery mechanism. Packets are sent by the sender into the network, and
they either make it to the receiver on time, arrive late, or are lost in transit.
So somehow we must insure that the packets associated with voice or video conferencing make it
though the network in a timely manner, without getting lost, and with no help from the transport
protocol. This is the challenge of supporting real-time applications. Quality of Service (QoS) is the
mechanism we deploy in our networks to give priority to the voice and video streams, to insure they
will be delivered correctly. We will explore the different QoS approaches and how to deploy QoS for
voice and video in this document.
3.0 What is Quality of Service (QoS)?
The term QoS has been used to describe many different ways of providing better service to some
types of traffic in a network environment. These can include priority queuing, application specific
routing, bandwidth management, traffic shaping and many others. We will be discussing priority
queuing since it is the most widely implemented QoS approach. Using priority queuing is not the
only approach that will work. Any approach that reliably delivers packets on time will support voice
or video conferencing. But since priority queuing is the most available QoS, we will describe here
how it works, and how to configure it to best support voice and video conferencing.
Enabling QoS in the network is only part of the problem. We will review here the four steps to
insuring QoS works correctly in your network as follows:
Network QoS Implementation
Bandwidth Demand and Bandwidth Availability
Queues are the primary contributors to packet loss and delay in a packet network. There are
queues at each output port in each router and switch in the network, and packets have to enter and
leave an output queue on each device through which it passes on its route. If queues are empty or
nearly empty, the packet enters and is quickly forwarded onto the output link.
Input Queue Output
Figure 3 - Packet Queue Diagram
If momentary traffic is heavy, queues fill up and packets are delayed waiting for all earlier packets in
the queue to be forwarded before it can be sent to the output link. If momentary traffic is too high,
the queue fills, and then packets are discarded, or lost.
A priority queuing mechanism provides additional queues at each switch or router output port,
dedicated to high priority traffic. In Figure 3 a simple 2-queue output port is diagrammed. Best effort
traffic is all queued in the lower queue, while high priority traffic, here colored green, is queued in the
upper queue. The queue policy determines how the queues will be emptied when they both have
A simple priority queue, sometimes called a low latency queue, is always emptied before any lower
priority queue is serviced. So in Figure 3, if the queue policy is simple priority, both green packets
are serviced first, and then the remaining packets in the best effort queue. If additional high-priority
packets arrive while the lower queue is being emptied, service is immediately switched to the high
A rate based queue behaves slightly differently. A rate based policy empties queues based on the
bandwidth that has been allocated to each queue. If queue 1 is allocated 40% of the available
bandwidth, and queue 2 is allocated 60% of the available bandwidth, then queue 2 is serviced 6/4
times as often as queue 1. Using a rate based policy, the queue is serviced often enough to keep
the allocated flow moving rapidly through the queue, but if excess traffic arrives, it will back up in the
queue while priority is given to the other queue. In our example in Figure 3 the green packets may or
may not get forwarded first, depending on the queue bandwidth allocations, and depending on how
much traffic has been previously emptied out of each queue.
3.1 Network QoS Implementation
The network must have a QoS mechanism that operates at each switch and router to prioritize real-
time traffic. A number of different mechanisms exist in modern networks including IntServ (RSVP),
DiffServ, IEEE 802.1p/Q, and IP Precedence. Additionally some enterprises rely on over
provisioning, which really means not using a QoS mechanism, but instead insuring there is plenty of
bandwidth. We’ll discuss in a minute why this is a risky strategy.
To gain the maximum benefit, QoS must work from end-to-end. All routers and switches between
the real-time sender and the real-time receiver must have a QoS mechanism available and enabled.
Partial solutions will prevent packet loss or delay on the links they serve, but to insure quality
delivery, all links should be served.
Let’s review the QoS mechanisms available to determine which is best suited to supporting voice
and video conferencing traffic.
In the discussion above about queues we said that if the queue is empty or nearly empty, that
incoming packets will be forwarded without delay. Over provisioning takes advantage of this
principle by providing more link bandwidth than necessary, so that queues will be empty most of the
time. Enterprises who pilot video conferencing on high bandwidth campus links often are very
pleased with the result without any further QoS implementation. Unfortunately this is a short sighted
approach, and will inevitably lead to problems later on.
Implementing real-time traffic without a true QoS mechanism is a game of chance. We saw earlier
the very bursty nature of data applications. An interesting characteristic of data applications is that
as many applications are aggregated together, intuition would tell us that the peaks will be smoothed
out. This turns out not to be true; data traffic keeps its bursty profile over many levels of aggregation
and over many different time scales. So a large burst of data traffic is just waiting to happen, even
on a 1 Gigabit or 10 Gigabit backbone link. When that burst occurs, it will cause delay or packet loss
for the voice and/or video conferencing streams. Data traffic tends to be heaviest during the work
day, which also coincides with when voice and video conferencing take place, again making the
chances for packet loss more likely.
Layer 2 and Layer 3 Quality of Service
QoS is implemented both at layer 3 and layer 2 in the protocol stack. Layer 3 QoS will be
recognized and handled by routers in the network. However, congestion also occurs in switch
output queues, and so level 2 QoS is also required. This is less true in the WAN network, where the
structure tends to be long links connecting one router to the next. However, in the Enterprise
network it is often necessary to have both a layer 2 and layer 3 QoS deployment.
Layer 3 QoS
There are three methodologies available in most networks for implementing QoS at layer 3, IntServ
(RSVP), DiffServ, and IP Precedence. Let’s look at the difference between these approaches.
Integrated Services (IntServ) is a very comprehensive approach to QoS defined by the IETF to
insure the proper treatment of high priority traffic in a converged IP network. IntServ, using the
RSVP protocol, sends out a request to all routers on the path for a given real-time stream, requests
the use of a priority queue and asks permission to use a certain amount of bandwidth. If each router
responds indicating that those resources are available then the path is enabled, and the stream is
given priority along that path. IntServ works well in environments where all routers support the
protocol, and where the number of real-time streams is limited. Because RSVP requires each router
to maintain information about each active real-time stream, network architects quickly realized that
this solution would not scale to large sizes well. Too many resources are consumed in each router.
Few enterprises today implement IntServ as their solution.
Differentiated Services (DiffServ) was developed as an alternative approach that would have better
scaling properties. Rather than specifying resources for each real-time stream, DiffServ allocates
resources for a class of traffic. All traffic allocated to that class is treated with the same policy, such
as being queued in a high priority queue. The key difference is that whereas the network may have
a bandwidth limit for this high priority class of traffic, the network is not managing the demand for this
class, only serving the traffic in this class as best it can. So DiffServ makes the job of the network
easier, but moves the problem of bandwidth management back to the application. We will discuss
bandwidth management in detail later in this document.
IP Precedence is a methodology created in the original IP specification. It is a much simpler
mechanism that gives precedence to IP packets marked as high priority. The bit positions in the IP
header formerly used for IP Precedence and Type of Service (TOS) have been reassigned for use
with DiffServ. There is no advantage to using IP Precedence over DiffServ unless routers are
deployed that do not support the DiffServ standard.
Most Wide Area Network (WAN) vendors today are using DiffServ marking to manage QoS in their
networks. WAN networks are inevitably a part of the enterprise network using voice or video
conferencing, so there is value in choosing an approach that is consistent across both the campus
and wide area networks. Careful selection of the DiffServ markings will insure that both the campus
network and the WAN vendors treat voice and video conferencing streams with the correct policies.
Layer 2 QoS
The predominant technology in use today for providing QoS at layer 2 is IEEE 802.1p. Switches
purchased in the last 7 years are very likely to have this capability built in. IEEE 802.1p functionality
is often coupled with IEEE 802.1Q, which is the VLAN functions. Both technologies use the same bit
field added to the Ethernet header, to specify a streams priority and VLAN association.
Switches implementing IEEE 802.1p use multiple output queues for each switch output port. Higher
priority traffic is assigned to a higher priority queue, and those queues are serviced before lower
priority queues, just as explained earlier for the routers.
Table 1 shows the mapping of Ethernet IEEE 802.1p priority mappings to the DiffServ codepoints. If
a campus network is using 802.1p, this translation needs to take place wherever level 3 QoS meets
the level 2 QoS such as at the core routers or the WAN router.
It is a common misconception that assigning traffic to VLANs completely separates it in the network,
and that this is a viable solution for voice and video traffic. VLAN assignment insures that traffic will
not be forwarded to portions of the network where those VLANs are not allowed, so traffic separation
occurs from a permissions point of view. However, priority is assigned separately. Often VLAN
assignment and priority assignment are coupled, e.g. any traffic assigned to the Red VLAN gets high
priority. If this is true, then traffic in the high priority VLAN will get precedence at the switches, but
Table 1 - DiffServ to Ethernet IEEE 802.1p
the two assignments (VLAN and priority) are not
necessarily coupled. If there are more than
DiffServ Code Point PPP Class one ‘high priority’ VLANs passing through the
(DSCP) Number same switch, they will contend for the same
output queue resources.
CS7, CS6 7
EF, CS5 6 VLANs are often used as a method for marking
traffic. If the endpoint itself is unable to mark
AF4x, CS4 5
traffic, or is not trusted to mark traffic correctly,
AF3x, CS3 4 the VLAN assignment can be used to indicate
this is high priority traffic. VLANs can be as-
AF2x, CS2 3
signed to traffic arriving on a specified physical
AF1x, CS1 2 port, so any traffic arriving from a particular
DE, CS0 0 system, such as a room video conferencing
system, can be assigned to a specific VLAN,
Ref: Nortel White Paper “Introduction to
and thus to a specific priority level in the net-
Quality of Service (QoS)”
work. This is discussed further in Section 3.2.
A number of technologies exist for wide area networks, and each has its own QoS approach. We
will discuss leased lines, Frame Relay, ATM, and MPLS.
Table 2 - DiffServ to PPP Priority Mapping
Leased lines are the simplest technology to
DiffServ Code Point PPP Class manage from a QoS perspective, but often are not
(DSCP) Number the best choice for topology or cost. If an enterprise
leases a T1 or T3 circuit between two facilities, than
the enterprise router on each end is in full control of
CS7, CS6, CS5 6 how traffic is scheduled onto that WAN connection,
AF4x, CS4 5 and the link behaves like any other link in the
Enterprise network. In this situation, whatever QoS
AF3x, CS3 4 approach is chosen for the enterprise network can
AF2x, CS2 3 be used on these WAN links as well.
AF1x, CS1 2 Table 2 shows the mapping from DiffServ Code
DE, CS0 1 Points to PPP class numbers. This mapping is
done on the router that attaches to each end of a
Ref: Nortel White Paper “Introduction to point-to-point link.
Quality of Service (QoS)”
Frame Relay is a well established technology, and
is used heavily by enterprises that require relatively low bandwidth connections. The user is
cautioned against using a Frame Relay service to carry real-time traffic, as they often have difficulty
maintaining sufficiently tight jitter specifications, and will usually not guarantee jitter within the
specifications required for good real-time transport.
Frame Relay services provide classes of service that can be used to prioritize one traffic type over
another. These classes of service work well to insure interactive applications like Telnet or Citrix get
precedence over email transfers and file backup. But they are not designed to provide the kind of
priority that real-time traffic requires. Furthermore, because frame relay services often have multiple
PVCs using the same physical connection, it is difficult to get true priority on a priority queue. The
router serving multiple PVCs creates a virtual port for each PVC, each having a high priority and
best effort queue. There is no communications, however, between these two virtual ports, even
though they are using the same physical port. Hence high priority traffic sitting in a queue on one
virtual port cannot override traffic in the best effort queue on the other virtual port. This means true
priority queuing is not happening, and leads to intermittent voice and video quality problems.
ATM (Asynchronous Transfer Mode) is a 1990s technology which was heavily deployed by wide
area networking vendors. ATM has built-in sophisticated QoS mechanisms that do a good job of
Much of the technology for measuring and policing bandwidth flows was developed by the ATM
Forum during the initial days of ATM deployment. These techniques are used today in DiffServ and
ATM service categories are not the same as DiffServ categories, but the DiffServ categories can be
mapped into ATM categories at the network boundary. Table 3 shows the mapping between DiffServ
code points and ATM service categories.
Enterprises using ATM within the
Table 3 - DiffServ to ATM QoS Mapping corporation can use this QoS
DiffServ Code Point (DSCP) ATM Service Category mechanism as well. Larger
enterprises have ATM backbones
CS7, CS6, CS5, EF CBR or rt-VBR
between major sites. ATM QoS will
AF4x, CS4, AF3x, CS3 rt-VBR give priority to classes higher on this
AF2x, CS2, AF1x, CS1 nrt-VBR
DE, CS0 UBR MPLS (Multi-Protocol Label
Ref: Nortel White Paper “Introduction to Quality of Service Switching) is the latest WAN
(QoS)” technology and appears to be the
current or future plan of most WAN
service providers. MPLS has many of the characteristics of ATM that allow services providers
control over how traffic flows in their networks. This allows providers flexibility in how they offer
services, and allows them to quickly adapt to new market needs for different services. Because
MPLS allows providers this flexibility, many are offering services that include quality of service
guarantees. This has led to the misconception that MPLS implies QoS, which it does not. Often a
service provider will have built a good QoS offering on top of their MPLS implementation, but the
details of this QoS implementation need to be understood to insure that real-time traffic gets the right
Service providers using MPLS create different classes of service by configuring their routers to either
recognize bits in the MPLS tag, or by prioritizing specific routes based on their individual labels. For
either method, there are a fixed number of classes, often eight or less. Thus there will again be
some mapping that needs to be done between DiffServ markings and the specific traffic classes
implemented by the MPLS core. Most MPLS-based WAN providers are using DiffServ to identify
traffic classes. Work with your proposed MPLS-based vendor to understand how they map and
support the different DiffServ markings into the classes of traffic they support, and understand the
forwarding behavior they specify for each class in their network.
Recommendation – DiffServ Markings
A working group of the IETF has issued a draft document recommending DiffServ markings for
specific types of traffic.
These recommendations are shown in Table 4. The intent of this document is to create consistency
between WAN vendors to eventually allow quality of service to be carried across multiple WAN
providers while maintaining similar forwarding behavior through each. Using these markings will
make an enterprise compatible with WAN vendors following the IETF recommendations.
Table 4 shows Telephony (VoIP) using the EF marking, and video conferencing uses AF41, AF42 or
AF43 markings. Telephony signaling is slotted in between these two, using the CS5 marking. Video
conferencing signaling is not explicitly called out. It should go below the level of the video and audio
streams, but not be considered ‘Standard’ (which is Best Effort), since people are waiting on the
Table 4 - IETF Recommended DSCP Markings results of the signaling
Service DSCP PHB Used Queuing AQM transaction. The High
Class Throughput Data category
(AF11, AF12 or AF13) works
Network CS6 RFC2474 Rate Yes well for video conferencing
Telephony EF RFC3246 Priority No
Signaling CS5 RFC2474 Rate No The third and fourth column of
Table 4 shows the forwarding
Multimedia AF41 RFC2597 Rate Yes
behavior recommended, and
Conferencing AF42 per
refers to the appropriate IETF
specification for details. Note
Multimedia AF31 RFC2597 Yes that Expedited Forwarding
Streaming AF32 Rate Per (EF) is recommended for
AF33 DSCP telephony, which is
OAM CS2 RFC2474 Rate Yes implemented with a priority
High AF11 RFC2597 Rate Yes queue. As discussed earlier,
Throughput AF12 Per the priority queue is always
Data AF13 DSCP emptied before other queues,
giving the lowest possible
Low Priority CS1 RFC3662 Rate Yes latency to that traffic. It is
Data suggested here that video
Standard DF (CS0) RFC2474 Rate Yes conferencing be implemented
+ other using a high priority Assured
Reference Internet Draft draft-ietf-tsvwg-diffserv-service-classes-02. Forwarding (AF) marking,
which is implemented with a
rate-based queue. This is appropriate for a converged network where both voice and video
conferencing are being implemented, because it insures that the voice traffic has priority over the
higher bandwidth and bigger packets of the video conferencing stream. A rate based queue will
properly forward video conferencing traffic as long as the queue rates are set to exceed the worst
case amount of traffic generated by the video conferencing streams. Management of this bandwidth
is very important, and will be discussed in detail later.
Recommendations on separating the voice versus the video portion of a video conferencing stream
vary. Some claim that there is no value in getting the voice component of a video conference
delivered earlier than the video portion, since the receiving unit must delay the voice to synchronize
it with the video signal. There is, however, value in giving audio better priority if it means it will get
less interference with other streams, and thus be delivered more reliably. A video conference with
poor visual quality can proceed as long as the audio remains clean, but without good audio, no
It is possible to use Expedited Forwarding (EF) for video conferencing streams if video conferencing
is the only real-time traffic in the network. Most enterprises are either making the transition to VoIP
or are considering it in the near future, so using EF for video is not a good long term strategy.
Active Queue Management (AQM)
The last column of Table 4 indicates whether Active Queue Management (AQM) is recommended.
Notice that for telephony, the answer is no. The predominant AQM mechanism is Random Early
Discard (RED). This algorithm is designed to help manage multiple TCP flows when the link is near
the congestion point. RED determines that the queue length is getting long, and then randomly
selects packets within the queue for discard. When TCP streams lose packets, they back down their
sending rate, thus reducing the congestion.
As we discussed earlier, real-time streams lose quality quickly when packets are lost, so inducing
packet loss is not to their advantage. The IETF recommendation in Table 4 recommends that AQM
be used on video conferencing streams, under the assumption that the video conferencing endpoints
know how to reduce their sending rate when packet loss occurs. Assuming that functionality exists,
video conferencing streams would change their video rates to a lower bandwidth when they detect
packet loss, and this would relieve congestion in the network, making all streams again able to
A better approach is to insure that this condition does not happen in the first place. For a video
conferencing environment where room-based video conferencing systems are the primary users of
the video conferencing bandwidth, the bandwidth should be predicted, agreed upon and scheduled.
Relying on the endpoints to negotiate reduced bandwidth means that there is a temporary
interruption in the quality of the video conference, followed by a reduction in the video quality. This
is not acceptable in a business level video conferencing environment.
If making ad hoc video calls using desktop video conferencing capabilities becomes popular in the
future, it may be necessary to create two classes of video conferencing traffic, and to manage the
bandwidth of each separately. Managing the bandwidth of ad hoc video conferencing users has to
be done on an averaging basis, similar to the way it is done for voice calls. This group of users then
may have to degrade their video quality when the network gets congested or receive a busy signal if
insufficient bandwidth is available to place the video call. Business class video conferencing should
be kept independent of this ad hoc class so that scheduled calls can be placed with the scheduled
bandwidth, and obtain consistent, quality results.
Classification is the task of determining which streams deserve high priority treatment, and
identifying them with a marking so that the network switches and routers will recognize them. All the
previous discussion about QoS implementations assumes we know which streams ought to get
preferential treatment, and which streams should just get best effort support. Classification is the job
of making this decision and marking streams accordingly.
Classification by Endpoints
In many implementations it is possible for the endpoints themselves (phones, video conferencing
endpoints, gateways or bridges) to identify the voice and video streams. The endpoint is the most
knowledgeable component in the network, because the endpoint contains and understands the
application. It is a simple matter for the endpoint to identify which streams contain voice or video
content, and which streams are data transfers or control traffic. Most voice and video endpoints
have a configuration option that allows the QoS marking to be specified, and that marking is then
applied to the high-priority streams.
However, the network may not trust the endpoints to properly mark their streams. A video
conferencing system may be properly identifying its traffic as voice, video, data or control, but a PC
can also emulate these same markings for non-priority traffic, and take advantage of the better
service. This is like driving in the commuter lane with only one person in the car.
Classification by the Network
Networks often avert this problem by classifying data streams themselves. This means the edge
switch or router must look into the packets of the data stream to determine which ones are high
priority and which are not, and mark them accordingly. A network will do this classification at the
edge or access router; distribution, core and WAN routers will then trust the markings established at
the edge. Classification can be done on the basis of:
• IP address (well known end-point)
• TCP or UDP port number
• Physical port
Often a combination of these parameters is used, perhaps in conjunction with the markings
established by the endpoint itself. For instance, if a video and/or audio bridge is identified by its IP
address, which is statically assigned, all UDP-based traffic can be marked as high priority traffic.
Since the endpoint is a well-known IP address which is difficult to spoof, and the endpoint sends
primarily real-time traffic streams, this classification rule will work well. Conversely, if desktop IP-
phones or video endpoints are being user installed throughout the enterprise, it is difficult to have an
up-to-date list of ‘approved’ high priority devices to use in this classification approach. Different
portions of the enterprise may operate with different levels of trust with respect to endpoints making
their own traffic classifications.
Enterprise Classification Policy
Each enterprise should develop a policy on how classification will be accomplished. The policy
should scale well as the deployment of phones or video endpoints increases. The policy needs to
reliably identify real-time streams without compromising the integrity of the network. The need to
manage who is using the high priority service will be discussed in more detail in the bandwidth
Note that the default behavior of routers, if not configured to accept an endpoint classification, is to
remark the incoming packets to the best effort category. Enabling QoS in the system and marking at
the endpoints is often not sufficient. Insure that edge routers are properly configured to either trust
the endpoints or to classify and mark packets themselves. It is useful to capture data on the
receiving end and determine if packets maintained their QoS markings throughout their journey
across the network.
4.0 Bandwidth Demand
Bandwidth use is an integral part of QoS. Sufficient bandwidth must be in place on each link to carry
the expected real-time traffic. So the first question is what is the expected traffic? It is important to
analyze expected demand so that proper bandwidth planning can be done to support video
conferencing on the network links.
Bandwidth demand analysis can be done in a number of different ways. If a video conferencing
service already exists in the organization, then there is some history available of how many calls are
placed during the day, the locations called, the duration of those calls and the call bandwidth. This
information can be compiled into a demand graph per link, and then the maximum values can be
obtained. Figure 4 is an example of such an analysis. Video conferencing calls were mapped onto a
network diagram to determine which WAN links were used for each call. The time of day, duration
and bandwidth of the calls were noted. For each 30 minute period of the day, the bandwidth
consumed by each call active on each link was summed up to show the total demand on that link for
that period. Figure 4 shows those results for one of the WAN links. This link had a maximum usage
of 6.5 Mbps for video conferencing.
A similar approach can be used for an enterprise that is currently transporting video conferencing
traffic over ISDN lines. The call history can be used to determine what the demand on the IP
network would have been if those calls were instead carried on the IP network.
If no existing call information is available, then call density and patterns must be estimated based on
expected usage. If video conferencing will primarily be taking place from video conferencing rooms,
then demand can be estimated by making some assumptions about room utilization. Call
destinations will have to be estimated by someone with knowledge about the business and likely call
patterns for the users.
Peak Bandwidth (Mbps) 7
Figure 4 - Video Conferencing Bandwidth Demand
Video conferencing is usually used for a scheduled meeting, with duration of at least 30 minutes,
and often more like an hour or an hour and a quarter. A heavy deployment of desktop video
conferencing systems may change this dynamic, as users begin to use their video systems more like
a telephone, for ad hoc meetings which start at random times, and have shorter durations.
Modeling telephone usage is done on a statistical basis using Erlang tables. Given the frequency
and duration of calls, the Erlang calculation determines the amount of available bandwidth required
to obtain a given level of bandwidth availability, i.e. in order to not have calls be blocked. More
information on Erlang tables and a calculator can be found at www.erlang.com.
Bridge (MCU) Bandwidth Demand
Key infrastructure components of the video conferencing system need special consideration. First
consider the video conferencing bridge (or MCU). If 20 video conferencing endpoints are engaged
in a conference call, all 20 endpoints have established a full duplex connection to the bridge. The
bridge network connection must be able to sustain the maximum number of endpoints that will be in
all simultaneous conference calls. Thus the bridge should be placed near the core of the network
where bandwidth is more plentiful. Furthermore, the bridge should be placed in the facility where the
highest percentage of conference call users reside to minimize the WAN traffic required to support
these conference calls.
Each client that connects to the bridge will have a traffic stream flowing from the client to the bridge
at the bandwidth negotiated for that video conference. If each client has negotiated a 384K
bandwidth call, and there are 20 clients, the bridge will be supporting 384K x 20 or 7.7Mbps of traffic.
When we add the 20% additional bandwidth required for IP packet overhead, this now comes to
Some video conferencing endpoints also support a built-in conferencing mode. If a video
conferencing endpoint is acting as a bridge for a small conference, there will be a proportionate
increase in the bandwidth to that client. A 4-person conference using one of the 4 clients as a bridge
will generate three full duplex streams to the client acting as a bridge. The other three clients will
see a single full-duplex stream.
Gateway Bandwidth Demand
A gateway is often used to connect an IP-based voice or video system to the PSTN. Calls being
placed from within the IP infrastructure pass through the gateway to an ISDN or POTS connection.
Gateways can be stand alone units, or are sometimes incorporated into a Bridge.
Bandwidth demand for the gateway should be calculated based on the number of simultaneous calls
being placed through the gateway to the PSTN. This calculation is often straight forward, because
there is a limited amount of PSTN bandwidth. If the gateway is incorporated into the bridge, this
bandwidth should be added to the bandwidth due to conferencing.
IP and ATM Overhead
Bandwidth calculations need to take into account the overhead of an IP-based transport. In an ISDN
environment, a 384Kbps video conferencing call will consume 384Kbps of the available transport
bandwidth. In an IP environment the same
Table 5 - IP Bandwidth Overhead 384Kbps bit stream is broken up into packets,
Type Ethernet/ PPP ATM and those packets carry the additional overhead
MPLS of an RTP header, a UDP header, an IP header
and a layer 2 transport header. This overhead
Voice G.711 36% 25% 40%
must be added to each call bandwidth to
Video 20% 18% 25% determine the real impact on the IP network.
Bandwidth overhead can be approximated using
the values shown in Table 5. Voice overhead for
lower bandwidth calls like G.729 is higher than the values shown for voice in this table. Header
compression should be considered when many low bandwidth voice calls are carried across WAN
trunks, to help minimize overhead.
The video in Table 5 is the overhead for the combined video and audio streams of a video
conference. For example, a 384K video conference consumes 384Kbps x 1.2 = 460Kbps of network
5.0 Available Bandwidth
Once the bandwidth demand has been calculated, an evaluation of existing network bandwidth and
utilization is required to determine if there are sufficient resources to support the new real-time load.
Each link of the network needs to have sufficient bandwidth to support the voice and video traffic
expected, plus the existing data applications that use those same connections.
Although this sounds like a daunting task, in practice it usually means evaluating the wide area
network links, the backbone connections of the bridge, and client connections where there may be
10Mbps Ethernet or shared Ethernet connections. Often much of the infrastructure of an enterprise
does not need detailed bandwidth analysis, just these key elements.
Client connections should all be upgraded to 100 Mbps full duplex if possible. If the video
conferencing endpoint does not support full-duplex operation, it is preferable to run at 100Mbps half-
duplex. If the endpoint supports full duplex, but does not support 100 Mbps, it is preferable to run at
10 Mbps full duplex. When video conferencing runs on a half-duplex link, such as older Ethernet
links using a hub, the video conferencing application consumes a larger portion of the available
bandwidth. A 10 Mbps full duplex Ethernet connection supports 10 Mbps between the client and the
network switch, and another 10 Mbps between the network switch and the client. If the client is
running at 384K, it consumes 460Kbps in each direction, or 4.6% of the available bandwidth. If the
same client is running on a half-duplex Ethernet, it consumes 9.2% of the available bandwidth.
There are two parameters to consider when evaluating the WAN links. First, the expected voice and
video (real-time) load should never exceed 35% of the link capacity. Priority-based QoS
mechanisms begin to lose their effectiveness at this level. Running with more than 35% high-priority
traffic means that the traffic starts to compete with itself, and reliable delivery is compromised.
The second parameter is the total bandwidth utilization of the link, including the real-time
components and the data components. It is straight forward to determine the bandwidth demand of
the real-time applications, but determining the needs of data applications is much more difficult.
Data applications, as we saw in Section 0, are very bursty, and when many of those applications are
aggregated on a link their profile is still very bursty. Data applications depend on bandwidth
overhead to get good performance. If the bandwidth of a link is limited to the average consumption
of the data applications, the applications themselves slow down, creating user frustration and
One method of determining the utilization required by existing data applications is to measure
current utilization during the busy hour of the day. Look at the utilization of each important link with
as fine a resolution as possible. Typical bandwidth monitoring tools average utilization over some
period of time (15 minutes, or an hour). This averaging smoothes out the utilization peaks, and
gives a false impression of how much bandwidth is needed for proper data application performance.
If monitoring can be done at a finer granularity (5 minutes or even one minute) more accurate results
Another useful metric is to talk to application users, and determine if they notice a reduction in
application performance during the busy hour of the day. This would indicate that bandwidth is
already a scarce commodity during those times. It may be useful to test those network links using a
synthetic traffic generator to simulate the expected video conferencing load, and determine the
impact on existing application performance. The testing should be done during the busy hour when
those applications are running at their peak load. If there is concern about impacting the business,
implement the tests slowly, adding additional synthetic bandwidth and monitoring application
If bandwidth is scarce, it is often valuable to evaluate the traffic flowing across critical links during the
busy hour to determine if some of that traffic does not support legitimate business purposes. Finding
and eliminating these unwanted traffic streams can often free up bandwidth.
Giving specific utilization values that are acceptable is difficult because each enterprise has a
different traffic mix and different needs. On the low end, 35% total utilization means good
performance for all applications. Seventy percent (70%) utilization is a high end limit in almost all
cases. But this leaves a wide range of choices.
Background tasks that are not time sensitive, such as email transfer, backups, downloads or
database synchronization, will work at higher load percentages. Applications where users are
waiting for an immediate response, such as HTTP-based applications, client-server applications or
keystroke applications like Citrix and Telnet, will be less tolerant. There is often value in extending
the QoS strategy to give these interactive applications a priority higher than the background tasks,
but lower than voice and video streams (see Table 4.)
6.0 Demand Management
If the network testing determines that there is insufficient bandwidth on critical links, the enterprise
has a few options to resolve the conflict:
• Bandwidth upgrade
• Reduce voice or video conferencing demand
• Compression / Application Acceleration Appliances
Bandwidth Upgrade - A bandwidth upgrade is always possible and may be the only solution if
insufficient bandwidth is available to carry the required voice or video conferencing load.
Limit Conferencing Demand - The second option is to limit the video conferencing demand. This
can be done in a number of ways. First, the bandwidth used by video conferencing calls can be
limited. Better video quality can be obtained at 1 Mbps or 512 Mbps, but quite good quality can be
obtained at 384Kbps, and even at 256Kbps. Use of the new H.264 video compression algorithm
enhances these lower bandwidth calls and provides better quality. So if the expectation of a remote
office was that they would be able to make calls at 512K, perhaps introduction of H.264 and a
reduction in call bandwidth to 384K or 256K will reduce demand sufficiently.
A second way to reduce demand is to manage call volume so that a limited number of calls can
occur simultaneously across each link. If a remote office has three video conferencing units, but the
bandwidth of the link can only support two simultaneous calls, a scheduling policy can be put in
place to insure that only two systems are being used concurrently. The simplest case of this policy
is to insure that the remote office only has the number of video conferencing endpoints that the link
Voice demand can be reduced by using a lower bandwidth codec (e.g. G.729) and by implementing
header compression on low bandwidth links.
The voice or video conferencing gatekeeper can also be used to help manage bandwidth utilization.
The gatekeeper can be assigned a maximum bandwidth available between pools of endpoints,
which relate to the topology of the network. The gatekeeper will then only allow calls across that link
up to the available real-time bandwidth allocated to that link. The bandwidth value given to the
gatekeeper is the maximum amount of real-time traffic allowed on that link, not the link capacity.
Once the link utilization reaches this maximum amount, the gatekeeper will refuse additional call
Overflow strategies can also be considered for those environments where a connection refusal is not
appropriate. If a data network has redundant paths to a remote office, application specific routing
can be employed to take advantage of the additional paths, to support a higher call volume. Another
approach is to have ISDN connections available, and to route overflow traffic through the ISDN
connections when necessary. Monitoring ISDN usage during busy hours will provide a simple
business case indicating when it is cost effective to add more IP bandwidth.
Compression – One more option is to compress the existing data traffic. A new class of data
appliances is appearing on the market that use various tricks to both reduce data traffic and increase
application performance simultaneously. These appliances use compression, caching, TCP
termination, transparent turns reduction and other techniques to accomplish their goals. There is a
bit of work to determine which approach best suits the data streams employed for each situation, but
these appliances can often make room on the link so that video conferencing or voice traffic can be
introduced without requiring a bandwidth upgrade.
Scheduling – The epitome of bandwidth management is to be able to schedule bandwidth at the
same time that a conference room and conference bridge ports are scheduled. Scheduling
bandwidth insures that the executive conference to be held next week will have the appropriate
bandwidth waiting for it when the call is set up, and that ad hoc voice or video conferencing users
have not pushed link bandwidth to the maximum just before the meeting is to start. In a centrally
managed video conferencing environment, this kind of bandwidth management is possible through a
manual process. Conferencing schedulers can insure that no more than the maximum number of
conferences are scheduled to use a particular network link during each half-hour period of the
conference, for example. Making this work requires conferencing schedulers to understand the
network topology as it applies to video conferencing connections.
In a more dynamic environment, with users making ad hoc calls without scheduling, a more
sophisticated approach is required. Polycom offers tools like Polycom Conference Suite (PCS) and
ReadiManager, which include endpoint, MCU port and bandwidth scheduling.
7.0 WAN Vendors and Technologies
Connecting enterprise locations is often done with the help of a Wide Area Network (WAN) service
provider. The service provider links become an integral part of the enterprise network, and of the
real-time traffic support. Thus it is important that we take a close look at how service providers
implement QoS on their access links and within their own core networks, and understand the impact
on the transport of the enterprise real-time streams.
7.1 QoS in the WAN Core
WAN service providers have a high speed core network, often using links at OC48 (2.4Gpbs) or
OC192 (9.7 Gbps) data rates. Jitter is minimal at these data rates because queues are emptied so
quickly. Some vendors claim that they can provide high quality transport for real-time traffic without
implementing QoS in their core networks. The reader is cautioned to think this through carefully
before deploying with one of these vendors.
Remember our discussion in section 0 about over provisioning? This is the strategy that these
vendors are implementing. By providing a very high bandwidth core, they hope to have minimal
packet loss and jitter, and thus be able to provide good quality without the overhead of implementing
a QoS algorithm. Often these vendors will provide a Service Level Agreement (SLA) that indicates
they meet specifications sufficient to support real-time traffic. We will discuss below how to read and
evaluate these SLAs.
There are WAN vendors today who are providing classes of service for different types of streams.
The reader is encouraged to seek them out and implement a WAN strategy using these service
providers, because they can give better guarantees for quality transport of real-time traffic.
7.2 QoS in the WAN Access Links
Of the many network links between two voice or video conferencing endpoints, the WAN access link
is likely to be the connection with the least available bandwidth. This means it is the link with the
highest probability of causing packet loss or jitter problems. Queuing problems occur where high
speed links are being switched through to a lower speed link, which is the case at each end of the
access link. Implementing QoS on this link is critical.
Prioritizing traffic entering the WAN is straightforward, because this traffic comes from the enterprise
LAN and thus is under control of the enterprise routers. If proper classification is done within the
LAN, the real-time traffic can be prioritized as it leaves the LAN and enters the WAN access link.
Prioritizing traffic as it leaves the WAN core and enters the access link has to be the responsibility of
the WAN vendor. If the WAN vendor is implementing QoS, this feature will be supported. If the
WAN vendor is not implementing QoS in the core, insure that at the very least they are implementing
QoS on traffic as it leaves the core and enters the access link. Also insure that the vendor is
carrying the QoS markings across the core so they are still available to traffic at this egress router.
A good test is to capture traffic as it enters the enterprise from an access link, and verify that QoS
markings are intact.
7.3 SLA Interpretation
The Service Level Agreement (SLA) provided by a WAN vendor will have specifications for
availability and performance as well as specifics of responsiveness when there are problems on the
link and how the enterprise will be compensated for those problems. We will concentrate here on
the performance specifications.
Performance specifications should include a value for latency, packet loss and jitter. Latency is
affected by the geographic distances involved with the WAN connection due to the limitations of the
speed of light. An SLA that includes connections from continent to continent will have longer latency
specifications than for an SLA that only includes domestic connections.
Check carefully to see if the SLA covers traffic flowing through the access links between the
enterprise and the WAN vendor. Some SLA specifications only guarantee traffic specifications
between the edge nodes of the WAN vendor’s network.
Packet loss and jitter specifications are often given as an average over some period of time. Take
careful note of this averaging period, it is often quite long, as much as a month. Like bandwidth
averaging, this averaging means the details of how the link is performing during important busy-hour
periods can be buried in the longer time-period specification. As an example, a packet loss
specification of 0.1% over a month period could mean:
Nights and Weekends loss = 0.02%
Work day loss (except busy hours) = 0.05%
Busy hour loss (2 hours/day) = 1.3%
So instead of getting the expected 0.1% packet loss during the important busy hour period, the
network is failing at 1.3% loss, causing significant video and voice impairment, and all within the SLA
specification. A more appropriate SLA specification would have performance guarantees over much
smaller periods of time, like an hour.
7.4 Enterprise to WAN QoS Handoff
Most WAN implementations today recognize the IETF DiffServ codepoint markings, and use those to
map traffic into specific traffic classes. It is important to align the enterprise QoS classes with the
WAN vendor classes, and to have consistency between different WAN vendors where more than
one vendor is employed within an enterprise. It is possible to remap QoS markings at the edge of
the WAN cloud so that enterprise markings can be properly aligned with WAN classes, but this adds
unnecessary complexity if it is not needed. If a QoS strategy is being developed, poll the current
and likely future WAN vendors to learn their QoS marking strategy, and align the enterprise strategy
7.5 Real-Time over VPNs or the Internet
Many small to medium sized enterprises today are taking advantage of Virtual Private Networks
(VPNs) to connect their geographically distributed offices. VPNs create an encrypted tunnel through
the public Internet. The advantage of a VPN is that the cost is often much less than a dedicated
connection. Enterprise VPNs come in two flavors, those that connect two offices through a single
WAN provider, and those that use the open Internet, so they may use more than one service
provider and their associated peering points.
Carrying real-time traffic through these VPNs is as risky as using over-provisioning for QoS. There
is usually no QoS capability offered in the VPN connection. Quality can often be reasonable in the
case where a single service provider is providing connectivity at both ends, but again no guarantees
about bandwidth, loss or jitter are available. Some enterprises use this approach anyway, because
the value of a voice call or video conference to an Asian manufacturing plant or a European
development center justifies the risk, and because the users can be tolerant of failures. If the quality
expectation is high, supporting management staff meetings, sales updates, presenting to clients or
other high visibility uses, than the risk of quality degradation and call failure may be too high to allow
Using the Internet for real-time traffic carries the same risks as a VPN, with less control. When
connecting to another party via the Internet, multiple carriers may be involved, and the user has no
control over how the call is routed. Hot-potato routing algorithms often insure that traffic flowing one
direction will take a different route than traffic flowing in the reverse direction. Educational and
research institutions have had some luck using the Internet where they have very high bandwidth
connections, but the risk of having a poor quality connection is high.
8.0 Network Verification
Testing the IP network for real-time support is the only way to get a real understanding of the state of
an IP network. A number of vendors offer testing tools that can be used to test the network, or a
consultant can be contracted to do a network audit. This step provides direct evidence of the ability
of the network to handle the bandwidth and timing requirements that real-time traffic imposes.
Synthetic test tools use a hardware or software agent, installed at locations around the network to
represent voice or video clients. These agents are then coordinated by a central console to conduct
real-time tests. Tests can be constructed to mimic the real-time traffic expected in the target
network, so voice calls and video conferencing calls of various quantities and bandwidths can be
Traffic flowing between these two agents stresses the network with the bandwidth consumption it
creates. If the consumption of this additional bandwidth impacts the performance of other business
applications, this can be noted and further bandwidth investigation started to solve those issues.
The receiving agent for each test stream also checks the packet stream for latency, packet loss and
jitter. These three key characteristics determine the network’s ability to support the timely delivery of
real-time data. Where test streams indicate poor real-time performance, diagnosis work can be
done to determine where packets are lost, or where jitter is introduced.
Network verification often finds both high level problems and lower level problems. High level
problems are ones that may have been missed during the design stage, such as enabling the
appropriate QoS, losing the classification markings at an edge router, or insufficient bandwidth on a
link. Lower level problems may include routers with an out-of-date software revision, an incorrect
access list definition, or insufficient memory or CPU resources. Poor LAN links, such as shared
10Mbps Ethernet connections or CAT 3 cables can also cause poor real-time performance. A
common issue is a poorly functioning Ethernet negotiation, where one end of the link resolves to a
half-duplex configuration and the other end resolves to a full-duplex configuration. Data traffic will
often work well across this mismatch, but real-time traffic
Table 6 - Real Time Test Parameters
Packet Loss < 0.1% Network verification should be done well ahead of real-time
traffic deployment, so that any issues uncovered during the
Packet Latency <= 100 ms verification process can be resolved before real-time traffic
Packet Jitter < 40 ms begins to flow.
Table 6 shows good target values for an enterprise network
supporting voice or video conferencing. Meeting these goals will insure quality voice and video
Packet Loss is often the most difficult parameter to meet. Many vendors’ products will indicate that
they can deliver good quality voice or video with much higher packet loss values than shown here.
Each of those approaches uses some kind of algorithm to mask the effect of lost information.
Whenever information is lost, the quality degrades, so the products have to make guesses as to
what the missing audio or video information was doing during the time represented by the missing
packet. Although these approaches can yield good results, the best approach is to not lose the
information in the first place. Because of the dynamic nature of the IP network, there will be plenty
of opportunities to test packet loss concealment algorithms when something goes wrong. Design to
make the network clean, and use packet loss concealment as a backup plan.
Latency affects the quality of an interactive voice or video conferencing call by slowing down the
response from the other party. After latency reaches 200 ms the delay effect is noticeable, and
parties have trouble interrupting each other, and maintaining the flow of a normal conversation. The
200 ms value represents the delay from the speaker to the listener, including all the delays of the
encoding, transmission and decoding of the signal. The network is only involved with the
transmission portion of this delay. Network latency should be kept below 100ms to insure that
speaker-to-listener latency stays below 200ms.
Latency is affected by congestion and by geographic distance. Congestion can be managed
through bandwidth management and QoS, but geographic distance and the speed of light are harder
to manage. Using a satellite incurs a long delay because of the distance signals have to travel up
and back to a geostationary satellite. In some global routing cases it is possible to get better routes.
The path traffic takes from Asia to Europe, for instance, often flows through the United States.
Finding carriers that will route this traffic by a more direct geographic route can lower the latency
impact. Enterprises not using satellites and working within a single continent will not experience
latency problems due to distance.
Jitter is the variation in packet delay as packets cross the network. Jitter is usually measured as
packet inter-arrival time, which isn’t quite the same thing, but works well for a packet stream that is
sent at a periodic interval like a voice stream. Jitter is primarily caused by queue delays. If a packet
passes through a nearly empty queue, its delay is short. If the next packet behind it finds the same
queue nearly full, it waits a long time before being forwarded. Jitter management is done by
managing queue depths. A shorter queue means packets won’t have to wait long, but it also means
bursts of traffic will cause packet loss sooner. So by managing jitter properly, it pushes the problem
back onto packet loss.
Jitter is managed by a receiving endpoint with a jitter buffer. This buffer holds an on-time packet for
some period of time before playing it. The depth of the jitter buffer determines the amount of time an
on-time packet is held. A late packet will be moved through the jitter buffer quickly, to bring it up to
the right ‘play’ time. Jitter is specified in Table 6 as 40 ms because some Polycom conferencing
systems have a 40 ms jitter buffer. This means packets arriving as much as 40 ms late can still be
played on-time, but packets arriving later than this will be discarded. The network needs to keep
jitter within this bound to prevent more packets being dropped, and the subsequent degradation of
voice or video quality.
9.0 Network Monitoring
Network monitoring is the ongoing version of network verification. Networks are very dynamic;
changes and additions are made to the network every day. The management tools used for data
networking are often insufficient to track and manage issues with real-time support, because they
are not measuring the right parameters, and are not measuring with sufficient granularity. To
properly maintain a network supporting real-time traffic, a real-time measurement tool needs to be
Network verification tools need the following characteristics to properly monitor the health of an IP
network carrying real-time traffic:
• Testing end-to-end across the network
• Testing packet loss, jitter and latency
• Storing historical data into a database for forensic analysis
• Thresholds available to cause alarms when quality degrades
Network monitoring tools built for supporting VoIP will also calculate and report Mean Opinion Score
(MOS), a measure of the quality of the voice call. A similar measure for the perceived quality of a
video conferencing image is being standardized by the ITU, and vendors will be including this
calculation in tools in the near future. This allows the tool to present a test result that directly
represents the quality of the call, giving IT managers a way to judge the effectiveness of their IT
Network monitoring can be done with purpose-built hardware and software tools. A number of
vendors supply test tools to support the testing of networks carrying real-time traffic (See Appendix
1). These tools place hardware or software probes around the network which generate small
amounts of synthetic traffic, and measure the ability of the network to properly deliver that traffic.
Some of these tools will also passively monitor real-time traffic streams to determine their quality.
An alternate approach is to take advantage of the voice or video conferencing endpoints involved in
real-time connections. Often these devices monitor the quality of the traffic delivered to them from
the far end, and record this information in a Call Detail Record (CDR.) Quality information may also
be available dynamically by reading MIB data from the endpoint, or opening a management
connection to the endpoint during a voice or video call. Long term collection of the CDR information
is useful for tracking the overall quality of the service being offered, and helps find correlations
between poor quality and the endpoints or geographies involved in those calls.
Often a combination of these two monitoring approaches provides the best and most timely
information to the network team about the ability of the network to carry real-time traffic.
Determining and implementing a monitoring strategy is important to managing real-time call quality.
10.0 Enterprise Service Level Agreement (SLA)
Introducing real-time traffic into the enterprise changes the way both the voice/video team and the
network team operates. A new set of expectations is suddenly applied to the network, and
traditional approaches no longer work. This change can cause difficulties between the teams, and
slow down the deployment of voice or video. It also interferes with the prompt resolution of
problems, as each team blames the other for the current issues.
Developing an Enterprise SLA is a way of bridging the gap of misunderstanding between the teams,
during the transition period. It helps both teams be successful while the organization learns how to
adjust to the new reality of real-time traffic on the IP network.
An Enterprise SLA is a written agreement between the voice/video team and the network team,
which defines the characteristics required of the IP-network to support real-time traffic. It is very
similar to the agreement written with an external WAN provider. This document specifies the
following parameters required of the network:
• Bandwidth – Expected real-time traffic load by link
• Latency – Maximum latency between any two video or voice endpoints
• Packet Loss – Maximum allowed packet loss during any portion of the business day
• Jitter – Maximum allowed packet jitter during any portion of the business working day
Within the specification of these items will be included the timeframes over which these
measurements are taken, and with what granularity the information is captured (hourly, quarter-
hourly, etc.) Specifications should be written so that quality can be maintained during the busiest
hours of the work day.
These four items are the most critical. Additional items in the SLA can include process definitions for
how users will report problems, and how the two organizations will communicate when the problems
are real-time related. Additionally the SLA needs to have a process for renegotiating these
parameters, especially bandwidth, as the real-time deployment grows.
The SLA document gives the voice/video team a network specification against which they can test.
If testing tools are in place to test this SLA, the voice/video team can quickly determine if a reported
problem is due to a network failure or an equipment failure. This eliminates the finger pointing, and
gets the problems resolved in a timely manner.
For the network team, the SLA document defines the resources required to support voice or video
conferencing. Transport over the IP network is often considered ‘free’, but the IT team knows that
real money is required to support new applications. The specific requirements defined in the SLA
document allow the network team to define the infrastructure support required to meet the
specification, and to request the appropriate resources from management to support it.
The following checklist is a summary of this document, and can be used to determine how well the
network is prepared for a voice/video implementation.
Identify real-time bandwidth demand by link
Identify data bandwidth demand by link (utilization)
Assess available bandwidth by link, and upgrade as necessary
Determine enterprise QoS classification approach
Determine enterprise QoS technology for the Campus networks (LAN)
Determine enterprise QoS technology for the WAN, and select WAN vendors
Verify WAN support for enterprise QoS approach through testing
Deploy QoS in all portions of the network supporting real-time traffic
Verify network support of real-time traffic with synthetic testing
Implement a bandwidth management methodology
Implement real-time network monitoring capability
Create, negotiate and sign an enterprise SLA
Deploying applications using real-time traffic, like voice over IP or video conferencing, creates a new
and different challenge for the IP-network team. A successful deployment requires careful attention
to the requirements of real-time traffic. If each of the steps outlined in this document are tackled and
then incorporated into the daily operations of the network, the enterprise can not only have a
successful deployment, but maintain a high quality service over the IP-network through the inevitable
changes in applications, locations, and the network itself.